Giorgio Incantalupo
2009-Mar-16 10:51 UTC
[asterisk-users] url in dial command: how does it work?
Hi, Does anybody knows where I can find some docs about how to make the URL parameter inside the Dial command work? I tried to make some tests with a sip phone without success: the sip debug shows no URL inside sip packets. :( Any hint appreciated. :) Thank you Giorgio
Use IAX :-) In principle chan_skype could also support it. T. On 16 Mar 2009, at 10:51, Giorgio Incantalupo wrote:> Hi, > > Does anybody knows where I can find some docs about how to make the > URL > parameter inside the Dial command work? I tried to make some tests > with > a sip phone without success: the sip debug shows no URL inside sip > packets. :( > Any hint appreciated. :) > > Thank you > > Giorgio > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-usersTim Panton - Web/VoIP consultant and implementor www.westhawk.co.uk -------------- next part -------------- A non-text attachment was scrubbed... Name: smime.p7s Type: application/pkcs7-signature Size: 2419 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20090316/83570fc2/attachment.bin
Giorgio Incantalupo
2009-Mar-16 13:04 UTC
[asterisk-users] url in dial command: how does it work?
Hi Tim, ok, but I think the big question is...what is the URL for? It seems I need a special device...but which? What kind of device do you use? Thanks. Giorgio Tim Panton wrote:> Use IAX :-) > > In principle chan_skype could also support it. > > T. > > On 16 Mar 2009, at 10:51, Giorgio Incantalupo wrote: > >> Hi, >> >> Does anybody knows where I can find some docs about how to make the URL >> parameter inside the Dial command work? I tried to make some tests with >> a sip phone without success: the sip debug shows no URL inside sip >> packets. :( >> Any hint appreciated. :) >> >> Thank you >> >> Giorgio >> >> _______________________________________________ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users > > Tim Panton - Web/VoIP consultant and implementor > www.westhawk.co.uk > > > > ------------------------------------------------------------------------ > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users
Giorgio Incantalupo
2009-Mar-16 13:41 UTC
[asterisk-users] url in dial command: how does it work?
Hi Tim, it seems that using trunks is the right way....is this what you meant? Tim Panton wrote:> Use IAX :-) > > In principle chan_skype could also support it. > > T. > > On 16 Mar 2009, at 10:51, Giorgio Incantalupo wrote: > >> Hi, >> >> Does anybody knows where I can find some docs about how to make the URL >> parameter inside the Dial command work? I tried to make some tests with >> a sip phone without success: the sip debug shows no URL inside sip >> packets. :( >> Any hint appreciated. :) >> >> Thank you >> >> Giorgio >> >> _______________________________________________ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users > > Tim Panton - Web/VoIP consultant and implementor > www.westhawk.co.uk > > > > ------------------------------------------------------------------------ > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users-- _________________________________________________ Giorgio Incantalupo, mailto:gincantalupo at fgasoftware.com FG&A srl - http://www.fgasoftware.com - Voice at Work - The Agile PBX http://www.voiceatwork.eu Tel: 02997663.14, Fax: 0291390172